kphone lief einwandfrei und plötzlich zerhackt es den ausgehenden Ruf

Hi,

kphone funktionierte erst. Die Einstellung hab ich mir hier abgeschrieben.
http://www.ip-phone-forum.de/showthread.php?t=70600
Und jetzt, ohne einen Grund kommt meine Stimme an der Gegenstelle nur noch zerhackt an. Ich gehe über Web.de. Ich kann aber nicht mehr genau nachvollziehen ob zwischem dem Funktionieren und dem Defekt ein Neustart gelegen hat. Aber ich glaub es lief wenigstens zwei Tage astrein.

Das Testen verschiedener Codecs fürte auch zu nichts. Ich poste hier mal auf gut Glück was kphone über die Konsole ausgibt wenn ich telefoniere.

----------------------------------------------------------------------------------------------------------------------------
Found 2 interfaces.
SipClient: Listening UDP on port: 5060
SipClient: Our address: 192.168.0.3
SipClient: STUN request
SipClient: Receiving message...
SipClient: STUN response
address_port: 33638
address: 80.143.74.25
KCallWidget: Switching calls...
CallAudio: listening for incomming RTP
UDPMessageSocket: Listening on 32920
DspOutRtp: STUN request
SipClient: Empfange Nachricht ...
SipClient: STUN response for RTP
CallAudio: Opening ALSA device for Output



----------<170 - 2730>--------------


CallAudio: Creating RTP->ALSA Diverter

SipClient: Sending: 12:44:40.414
--------------------------------
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 80.143.74.25:33638;branch=z9hG4bK4D213DD8
CSeq: 2900 INVITE
To: <sip:[email protected]>
Content-Type: application/sdp
From: "Michael Holz" <sip:[email protected]>;tag=6154D669
Call-ID: [email protected]
Subject: sip:[email protected]
Content-Length: 228
User-Agent: kphone/4.2
Contact: "Michael Holz" <sip:[email protected]:33638;transport=udp>

v=0
o=username 0 0 IN IP4 80.143.74.25
s=The Funky Flow
c=IN IP4 80.143.74.25
t=0 0
m=audio 33127 RTP/AVP 0 97 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30

SipClient: Sending to 'sip.web.de:5060'
SipClient: Receiving message...

SipClient: Received: 12:44:40.583
---------------------------------
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 80.143.74.25:33638;branch=z9hG4bK4D213DD8;rport=33638
CSeq: 2900 INVITE
To: <sip:[email protected]>;tag=f43d8ce4e4130f38b15965d9884f209e.9634
From: "Michael Holz" <sip:[email protected]>;tag=6154D669
Call-ID: [email protected]
Proxy-Authenticate: Digest realm="web.de", nonce="4416ad5f58fb47d043861ab78b7ef9
f420df4925"
Server: Sip EXpress router (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 sip-ha.web.de:5060 "Noisy feedback tells: pid=23142 req_src_ip=80.
143.74.25 req_src_port=33638 in_uri=sip:[email protected] out_uri=sip:02572
[email protected] via_cnt==1"


SipCall: Incoming response
SipTransaction: Incoming Response

SipClient: Sending: 12:44:40.583
--------------------------------
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 80.143.74.25:33638;branch=z9hG4bK4D213DD8
CSeq: 2900 ACK
To: <sip:[email protected]>;tag=f43d8ce4e4130f38b15965d9884f209e.9634
From: "Michael Holz" <sip:[email protected]>;tag=6154D669
Call-ID: [email protected]
Content-Length: 0
User-Agent: kphone/4.2
Contact: "Michael Holz" <sip:[email protected]:33638;transport=udp>


SipClient: Sending to 'sip.web.de:5060'
SipCallMember: localStatusUpdated: 407
WL: SipProtocol: HA1=1c9f70a6aa2b8ccb6719ba8628d2a976 (mcgutschy:web.de)
SipProtocol: Digest calculated.

SipClient: Sending: 12:44:40.646
--------------------------------
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 80.143.74.25:33638;branch=z9hG4bK4328A13B
CSeq: 2901 INVITE
To: <sip:[email protected]>
Proxy-Authorization: Digest username="mcgutschy", realm="web.de", nonce="4416ad5
f58fb47d043861ab78b7ef9f420df4925", uri="sip:[email protected]", cnonce="ab
cdefghi", nc=00000001, response="38dcf9f7fa28cf26d1376a2b1412ee4f", opaque="", a
lgorithm="MD5"
Content-Type: application/sdp
From: "Michael Holz" <sip:[email protected]>;tag=6154D669
Call-ID: [email protected]
Subject: sip:[email protected]
Content-Length: 228
User-Agent: kphone/4.2
Contact: "Michael Holz" <sip:[email protected]:33638;transport=udp>

v=0
o=username 0 0 IN IP4 80.143.74.25
s=The Funky Flow
c=IN IP4 80.143.74.25
t=0 0
m=audio 33127 RTP/AVP 0 97 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30

SipClient: Sending to 'sip.web.de:5060'
SipClient: Receiving message...

SipClient: Received: 12:44:40.859
---------------------------------
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 80.143.74.25:33638;branch=z9hG4bK4328A13B;rport=33638
CSeq: 2901 INVITE
To: <sip:[email protected]>
From: "Michael Holz" <sip:[email protected]>;tag=6154D669
Call-ID: [email protected]
Server: Sip EXpress router (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 sip-ha.web.de:5060 "Noisy feedback tells: pid=23135 req_src_ip=80.
143.74.25 req_src_port=33638 in_uri=sip:[email protected] out_uri=sip:02572
[email protected] via_cnt==1"


SipCall: Incoming response
SipTransaction: Incoming Response
SipCallMember: localStatusUpdated: 100
SipClient: Receiving message...

SipClient: Received: 12:44:40.927
---------------------------------
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 80.143.74.25:33638;rport=33638;branch=z9hG4bK4328A13B
From: "Michael Holz" <sip:[email protected]>;tag=6154D669
To: <sip:[email protected]>;tag=as30019196
Call-ID: [email protected]
CSeq: 2901 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 4820 4820 IN IP4 217.72.200.72
s=session
c=IN IP4 217.72.200.72
t=0 0
m=audio 10636 RTP/AVP 8 0 111 3 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=silenceSuppff - - - -

SipCall: Incoming response
SipTransaction: Incoming Response
SipCallMember: localStatusUpdated: 183
CallAudio: Using G711a for output
CallAudio: Sende an die Gegenstelle 217.72.200.72:10636
CallAudio: Opening ALSA device for Input



----------<170 - 2730>--------------


CallAudio: Creating ALSA->RTP Diverter
SipClient: Receiving message...

SipClient: Received: 12:44:42.967
---------------------------------
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 80.143.74.25:33638;rport=33638;branch=z9hG4bK4328A13B
From: "Michael Holz" <sip:[email protected]>;tag=6154D669
To: <sip:[email protected]>;tag=as30019196
Call-ID: [email protected]
CSeq: 2901 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
Content-Length: 0


SipCall: Incoming response
SipTransaction: Incoming Response
SipCallMember: localStatusUpdated: 180
CallAudio: Using G711a for output
SipClient: Receiving message...

SipClient: Received: 12:44:47.391
---------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.143.74.25:33638;rport=33638;branch=z9hG4bK4328A13B
Record-Route: <sip:217.72.200.89;ftag=6154D669;lr=on>
From: "Michael Holz" <sip:[email protected]>;tag=6154D669
To: <sip:[email protected]>;tag=as30019196
Call-ID: [email protected]
CSeq: 2901 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 4820 4821 IN IP4 217.72.200.72
s=session
c=IN IP4 217.72.200.72
t=0 0
m=audio 10636 RTP/AVP 8 0 111 3 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=silenceSuppff - - - -

SipCall: Incoming response
SipCall: Checking for Contact and Record-Route
SipCall: Setting Contact for this Call Member
SipTransaction: Incoming Response

SipClient: Sending: 12:44:47.392
--------------------------------
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 80.143.74.25:33638;branch=z9hG4bK4328A13B
CSeq: 2901 ACK
To: <sip:[email protected]>;tag=as30019196
From: "Michael Holz" <sip:[email protected]>;tag=6154D669
Call-ID: [email protected]
Route: <sip:217.72.200.89;ftag=6154D669;lr=on>
Content-Length: 0
User-Agent: kphone/4.2
Contact: "Michael Holz" <sip:[email protected]:33638;transport=udp>


SipClient: Sending to 'sip.web.de:5060'
SipCallMember: localStatusUpdated: 200
CallAudio: Using G711a for output
dtmfsenderTimeout
SipClient: Receiving message...

SipClient: Received: 12:45:10.273
---------------------------------
BYE sip:[email protected]:33638 SIP/2.0
Record-Route: <sip:217.72.200.89;ftag=as30019196;lr=on>
Via: SIP/2.0/UDP 217.72.200.89;branch=z9hG4bKd9df.b50a6c17.0
Via: SIP/2.0/UDP 217.72.200.72:5060;branch=z9hG4bK287036f3;rport=5060
From: <sip:[email protected]>;tag=as30019196
To: "Michael Holz" <sip:[email protected]>;tag=6154D669
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 16
Content-Length: 0
P-hint: rr-enforced


SipCall: Incoming request
SipCall: New transaction created
SipTransaction: Incoming Request
SipClient: Sending UDP Response
SipClient: Sending to '217.72.200.89' port 5060

SipClient: Sending: 12:45:10.273
--------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.72.200.89;branch=z9hG4bKd9df.b50a6c17.0
Via: SIP/2.0/UDP 217.72.200.72:5060;rport=5060;branch=z9hG4bK287036f3
From: <sip:[email protected]>;tag=as30019196
CSeq: 102 BYE
Call-ID: [email protected]
To: "Michael Holz" <sip:[email protected]>;tag=6154D669
Content-Length: 0
User-Agent: kphone/4.2
Contact: "Michael Holz" <sip:[email protected]:33638;transport=udp>
Record-Route: <sip:217.72.200.89;ftag=as30019196;lr=on>


KCallWidget: Starting force disconnect...
SipClient: STUN request
SipClient: Receiving message...
SipClient: STUN response
address_port: 33638
address: 80.143.74.25
[Gutschy@localhost ~]$

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